[asterisk-bugs] [Asterisk 0010356]: chan_ooh323 don't work correctly

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10356 
====================================================================== 
Reported By:                malufrj
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10356
Category:                   Addons/chan_ooh323
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:            1.4.8  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-01-2007 10:04 CDT
Last Modified:              08-01-2007 10:07 CDT
====================================================================== 
Summary:                    chan_ooh323 don't work correctly
Description: 
Hi people,
I wanna work with Asterisk 1.4, and need work with 2 channels, SIP and
H.323. 
Because I have one Proxy SIP, OpenSER, and other Proxy H.323, GnuGK. 
My tests with chan_h323 wasn't good and  I compiled chan_ooh323 in
replace. 
But until now, I still don´t get one perfect call. The RTP traffic don't
pass in the two directions or the call is broken. 
When I call H.323 to SIP, my sip softphone ringing and my h.323 terminal
show the message "XXXX is ringing", but when I answer, the call in SIP stay
active while the call in H.323 is broken (hangup). 
When I call SIP to H.323, the call is established but the RTP traffic just
work in H.323 to SIP direction while in the other direction become mute,
i.e., without RTP traffic. 
I am sniffering the messages SIP and H.323 in the clients and in the PROXY
to help us.
My software version is: Asterisk 1.4.8 and Asterisk-addons 1.4.2
thanks in advanced,
Thiago Maluf.
====================================================================== 

---------------------------------------------------------------------- 
 malufrj - 08-01-07 10:07  
---------------------------------------------------------------------- 
I uploaded three files:
asterisk_sip-h323_20070801.log : it's my Asterisk Console while I mad the
calls.
h323_log : Log created by OOH323 Channel
ooh323.conf : my OOh323 conf file
thanks. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-01-07 10:07  malufrj        Note Added: 0068205                          
======================================================================